Hi All,
I am using "asterisk-11.12.0" version and I am trying to setup secure call
(TLS + SRTP) between two extensions and while making a call, I got
following error
*CLI> == Using SIP RTP CoS mark 5
-- Executing [6004@from-office:1] Dial("SIP/6003-00000000",
"SIP/6004,20") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/6004
SSL certificate ok
== Problem setting up ssl connection: error:14094410:SSL
routines:SSL3_READ_BYTES:sslv3 alert handshake failure
[Nov 2 21:20:05] WARNING[3571]: tcptls.c:673 handle_tcptls_connection:
FILE * open failed!
I followed instruction given in "
https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial";, but no
luck.
I googled around the issue and found solution mentioned by Patrick (
https://www.mail-archive.com/asterisk-users@lists.digium.com/msg274038.html)
Did anyone has tried this solution and found it is working? I tried to
create certificates with keyUsage/extendedKeyUsage, but it is not working.
I have one more query - When the SIP user agents are able to register
successfully with TLS, why more handshake is required while making a call?
Can't Asterisk use existing TLS connection with Leg B to forward INVITE
request? Could anyone please educate me on the same? I am little confused
here.
Thanks in advance.
--
Atul Thosar
--
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